1. Field of the Invention
This invention relates to a noise reducing apparatus of the type which reduces noise by generating a signal which is equal in amplitude but reverse in phase to the noise to reduce the noise, and more particularly to a noise reducing apparatus of the type mentioned which is suitably used for reduction of periodical noise and also to a noise reducing apparatus which reduces noise in a room of an automobile.
2. Description of the Prior Art
In a room of an automobile, noise is generated by rotation of an engine, and in an air conditioning system, noise is generated by rotation of a fan or a compressor. Such noise often causes people there to have a disagreeable feeling.
One of methods of reducing such noise is disclosed, for example, in Japanese Patent Laid-Open Application No. Heisel 1-178846 wherein s signal which is equal In amplitude but reverse in phase to noise is generated using an adaptive filter to cancel the noise.
An exemplary one of conventional noise reducing apparatus of the type just described is shown in FIG. 9. Referring to FIG. 9, the noise reducing apparatus includes a pickup circuit 11 for picking up noise from a noise source 10, a pair of analog to digital converters (A/D) 12 and 16, a digital to analog converter (D/A) 13, a loudspeaker 14, an adaptive filter 7, a transfer characteristic compensation section 8, and a tap value updating section 9 for updating tap values of the adaptive filter 9.
A microphone 15 is installed at a location where it is desired to reduce noise.
The adaptive filter 7 corrects a signal picked up by the pickup circuit 11 for those portions which are different from noise which is generated from the noise source 10 and inputted to the microphone 15 so that a signal issued from the loudspeaker 14 and coming to the microphone 15 may be equal in amplitude but reverse in phase to the noise from the noise source 10.
The adaptive filter 7 is constituted from a digital filter formed from a tapped delay line as described in detail below with reference to FIG. 10. In particular, the adaptive filter 7 can transform a signal of any waveform by the Fourier transform to decompose the signal into a frequency spectrum. Further, if the frequency spectrum is the same, then a same waveform can be obtained by the Fourier transform. Accordingly, the adaptive filter 7 controls a passing spectrum so that the spectrum of a signal picked up by the pickup circuit 11 may be the same as the spectrum of a noise signal from the noise source 10 received by the microphone 15.
The tap value updating section 9 updates the tap values of the adaptive filter 7 to make a filter characteristic so that the passing spectrum may be the same as the spectrum of the noise signal.
The transfer characteristic compensation section generates a compensation signal to compensate, since a signal generated by the adaptive filter 7 is influenced by a time delay and a frequency band limitation before it comes to the microphone 15 by way of the digital to analog converter 13 and the loudspeaker 14, for the transfer characteristic so that the signal at the input to the microphone 15 may be equal in amplitude but reverse in phase to the signal from the noise source 10.
Also the transfer characteristic compensation section 8 may be constructed from a digital filter formed from a tapped delay line. A detailed construction of the transfer characteristic compensation section 8 is shown in FIG. 11. Referring to FIG. 11, the transfer characteristic compensation section 8 shown includes a plurality of delay elements 80-1 to 80-J each of which delays an input signal by a time equal to a sampling interval of sampling pulses inputted to the analog to digital converters 12 and 16. An output value of each of the delay elements 80-1 to 80-J is multiplied by a tap value by a corresponding one of tap value multipliers 81-0 to 81-J.
Thus, where the output value of the analog to digital converter 12 when t=t.sub.n tn is represented by x(n) and the output value subsequently when t=t.sub.n+1 is represented by x(n+1) and besides a sum ##EQU1## a compensation signal C(n) from the transfer characteristic compensation section 8 outputted from an adder 82 is given by ##EQU2##
Referring now to FIG. 10, the adaptive filter 7 includes delay elements 70-1 to 70-Z, tap value circuits 71-0 to 71-Z and an adder 72. The delay elements 70-1 to 70-Z successively delay an output signal of the analog to digital converter 12 each by a time equal to the production interval of sampling pulses.
Accordingly, the output y(n) of the adaptive filter 7 is given by ##EQU3## and the output (y) is converted into an analog signal by the digital to analog converter 13 (FIG. 7) and then sent out from the loudspeaker 14.
The tap values W.sub.0 (n) to W.sub.z (n) of the adaptive filter 7 are updated each time a sampling pulse is generated. Such updating of the tap values is performed by the tap value updating section 9.
The tap value updating section 9 includes three stages of multipliers 90. 91 and 92 and a stage of adders 93.
An output signal C(n) of the transfer characteristic compensation section 8 is successively inputted to the delay elements 90-1 to 90-Z. by each of which it is delayed by a time equal to the production interval of sampling pulses.
Meanwhile, the multiplier 91 multiplies, by .alpha., a signal e(n) obtained by conversion of an output e(t) of the microphone 15 into a digital value by the analog to digital converter 16. The value g is determined depending upon the loop characteristic of the adaptive control system.
The tap value updating section 9 then performs calculation of updated values W(n+1) of the tap values of the adaptive filter 7. In order to facilitate description, updating of the tap value W.sub.0 (n) of the tap 71-0 to Wo (n+1) will be described as an example.
At the multiplier 92-0, multiplication between an output of the multiplier 91 and an output value C(n) from the transfer characteristic compensation section 8 is performed. The adder 93-0 subtracts an output value of the multiplier 92-0 from the tap value W.sub.0 (n) at a sampling time of t=t.sub.n and updates the tap value with a result of the subtraction as a tap value W.sub.0 (n+1) at a next sampling time of t=t.sub.n+1. In particular, updating of the tap value given by EQU W.sub.0 (n+1)=W.sub.0 (n)-.alpha. C(n) e(n) . . . (3)
is performed. Further, also for any other tap W.sub.i, updating of the tap value given by EQU W.sub.1 (n+1)=W.sub.i (n)-.alpha. C(n-1) e(n) . . . (4)
is performed.
Since the tap values are updated in such a manner as described above, sound waves sent out from the loudspeaker 14 are equal in amplitude but reverse in phase to noise from the noise source 10 at the input of the microphone 15, and consequently, noise in the proximity of the microphone 15 is cancelled or reduced.
As described above, the conventional noise reducing apparatus is constructed such that a noise signal picked up from a noise source is passed through an adaptive filter to generate a signal which is equal in amplitude but reverse in phase to the noise to reduce the noise.
To this end, the adaptive filter must perform a number of multiplications equal to the number of taps, and a number of multiplications and additions equal to the number of taps are required for updating of the tap values.
If the multiplications and the additions are performed with individual multipliers and adders constructed therefor, the construction of the apparatus is complicated very much, and therefore, they are normally performed by processing by means of a processor. However, a high speed processor is required in order for processing of multiplications and additions corresponding to a very great number of taps as described above to be performed within a time of an interval between sampling pulses, and this makes the cost of the apparatus high.
FIG. 12 shows an exemplary one of conventional noise reducing apparatus of the type described above which is specifically applied to reduce noise in a room of an automobile. Referring to FIG. 12, the noise reducing apparatus shown includes a microphone 101 installed at a location where it is desired to reduce noise, a noise signal pickup 104 for picking up a noise signal, an adaptive filter 105, and a loudspeaker 108.
The adaptive filter 105 corrects a signal picked up by the noise signal pickup 104 for portions which are different from noise inputted to the microphone 101 so that a signal developed from the loudspeaker 108 in accordance with the corrected signal and coming to the microphone 101 may be equal in amplitude but reverse in phase to the noise.
When the signal corrected by the adaptive filter 105 and developed from the loudspeaker 108 is, at the location where the microphone 101 is disposed, not a signal which is equal in amplitude but reverse in phase to the noise, a signal corresponding to a difference between the signal and the noise appears in the output of the microphone 101 and fed back to the adaptive filter 105.
The adaptive filter 105 is constituted from a digital filter formed from a tapped delay line not shown, and a transfer characteristic compensation section not shown.
As described above, a signal of any waveform can be decomposed into a frequency spectrum by the Fourier transform of the same. Further, if the frequency spectrum is the same, then the same waveform can be obtained by the inverse Fourier transform. Accordingly, the tap values of the digital filter are varied in response to a feedback signal from the microphone 101 to make a filter characteristic so that the same spectrum as the spectrum of noise may be obtained.
Meanwhile, the transfer characteristic compensation section compensates for an influence of a time delay and a frequency band limitation while a signal generated by the digital filter comes to the microphone 101 so that a signal equal in amplitude but reverse in phase to the noise may be obtained at the input to the microphone 101.
In this manner, the conventional noise reducing apparatus for an automobile is constituted from the microphone 101, the noise pickup 104, the adaptive filter 105 and the loudspeaker 108.
The loudspeaker 108 develops a signal having the same spectrum as the noise. Since the frequency spectrum of such noise includes noise components from a low frequency component to a high frequency component, in order for low frequency components to be developed efficiently, the loudspeaker must be sufficiently large.
Therefore, the loudspeaker is installed, for example, below a seat of an automobile as seen in FIG. 12 also taking the good appearance of the room of the automobile into consideration.
On the other hand, the microphone 101 is installed at a location where noise is desired to be reduced. In particular, the microphone 101 is installed at a location near the ears of a driver or a passenger of the automobile such as, for example, on the ceiling above the seat as seen in FIG. 12.
Where the loudspeaker and the microphone are installed in a spaced relationship from each other in this manner, the noise reducing effect is high for noise of low frequencies, but is as low as zero for noise of high frequencies as seen from FIG. 13.
It is considered that this arises from the fact that, when sound waves propagate in a closed space of a room of a vehicle, the transfer characteristic is fluctuated by such a cause that sound waves of a high frequency propagate with a greater number of reflections than sound waves of a low frequency. In other words, it is considered that the transfer characteristic set to the adaptive filter and the actual transfer characteristic are made different from each other for a high frequency signal by the variation in ratio at which the driver and/or passenger or passengers occupy in the space (whether they are fat or slim) and the variation of the sitting positions of them.